More buffer overflow checks from Corydon76.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -626,7 +626,6 @@ static char *ditch_braces(char *tmp)
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{
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char *c = tmp;
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char *n;
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c = tmp;
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if ((n = strchr(tmp, '<')) ) {
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c = n + 1;
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while(*c && *c != '>') c++;
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@ -2528,13 +2527,13 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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if (codec > -1) {
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snprintf(costr, sizeof(costr), " %d", codec);
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if (cur->codec < AST_FORMAT_MAX_AUDIO) {
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strcat(m, costr);
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strncat(m, costr, sizeof(m) - strlen(m));
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snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, cur->codec));
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strcat(a, costr);
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strncat(a, costr, sizeof(a));
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} else {
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strcat(m2, costr);
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strncat(m2, costr, sizeof(m2) - strlen(m2));
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snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, cur->codec));
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strcat(a2, costr);
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strncat(a2, costr, sizeof(a2));
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}
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}
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}
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@ -2550,13 +2549,13 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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if (codec > -1) {
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snprintf(costr, sizeof(costr), " %d", codec);
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if (x < AST_FORMAT_MAX_AUDIO) {
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strcat(m, costr);
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strncat(m, costr, sizeof(m) - strlen(m));
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snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
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strcat(a, costr);
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strncat(a, costr, sizeof(m) - strlen(a));
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} else {
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strcat(m2, costr);
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strncat(m2, costr, sizeof(m2) - strlen(m2));
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snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
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strcat(a2, costr);
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strncat(a2, costr, sizeof(a2) - strlen(a2));
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}
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}
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}
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@ -2568,20 +2567,24 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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codec = ast_rtp_lookup_code(p->rtp, 0, x);
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if (codec > -1) {
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snprintf(costr, sizeof(costr), " %d", codec);
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strcat(m, costr);
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strncat(m, costr, sizeof(m) - strlen(m));
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snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x));
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strcat(a, costr);
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strncat(a, costr, sizeof(a) - strlen(a));
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if (x == AST_RTP_DTMF) {
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/* Indicate we support DTMF... Not sure about 16, but MSN supports it so dang it, we will too... */
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snprintf(costr, sizeof costr, "a=fmtp:%d 0-16\r\n",
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codec);
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strcat(a, costr);
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strncat(a, costr, sizeof(a) - strlen(a));
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}
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}
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}
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}
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strcat(m, "\r\n");
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strcat(m2, "\r\n");
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if (strlen(m) < sizeof(m) - 2)
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strcat(m, "\r\n");
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if (strlen(m2) < sizeof(m2) - 2)
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strcat(m2, "\r\n");
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if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2)))
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ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
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len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a);
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if (p->vrtp)
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len += strlen(m2) + strlen(a2);
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@ -2925,6 +2928,8 @@ static int transmit_state_notify(struct sip_pvt *p, int state, int full)
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t = tmp + strlen(tmp);
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sprintf(t, "</dialog>\n</dialog-info>\n");
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}
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if (t > tmp + sizeof(tmp))
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ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n");
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snprintf(clen, sizeof(clen), "%d", strlen(tmp));
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add_header(&req, "Content-Length", clen);
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@ -4513,6 +4518,8 @@ static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header
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if (c)
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c++;
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}
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if (strlen(tmp) >= sizeof(tmp))
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ast_log(LOG_WARNING, "Buffer overflow detected! Please file a bug.\n");
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/* copy realm and nonce for later authorization of CANCELs and BYEs */
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strncpy(p->realm, realm, sizeof(p->realm)-1);
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